date:20050818

On Thursday 18 August 2005 12:13, James Fogg wrote:
> The interface is a non-Digium single port FXO card (modem) with R13 &
> R19 removed to mimic the Digium card. The software is the current
> release of [EMAIL PROTECTED] from ISO format (self booting CD that does an
> OS install and Asterisk compile automagically).

Make sure your PBX/KSU is giving a disconnect indication that asterisk 
understands (battery drop or battery reversal).  If not, there's not a damn 
thing you can do.

-A.
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[Asterisk-Users] Re: Which AGI Development Software is fastest on Asterisk?

2005-08-18 Thread Matt King


Hello,


I'm looking to develop some custom AGI that will be MySQL intensive.  It
appears Asterisk supports many different development environments.  Which
would be best suited for Asterisk and MySQL?


   It's generally fastest to use FastAGI (over TCP/IP), rather than 
regular AGI as this means the OS isn't starting a new process for each 
call (just like it's faster to use PHP or Servlets rather than 
old-school CGI for serving web pages).  This also means you can run your 
AGI application on a different server, if you want to, so as not to 
compromise Asterisk performance.


   If you know Java, you could try OrderlyCalls at 
http://orderlycalls.sourceforge.net (disclaimer - written by me!) which 
has full FastAGI and Manager support, reusable object pooling, and can 
be run inside Tomcat to build integrated web and telephony applications, 
though there are other packages out there, including Asterisk-Java ( 
http://asterisk-java.sourceforge.net - written by Stefan Reuter).


   Hope this helps,

  Matt King, Orderly Software
  http://www.orderlyq.com - probably the most advanced queue system 
in the world!


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Re: [Asterisk-Users] libpri mwi functionality?

On Thursday 18 August 2005 18:15, Gary Reuter wrote:
> I really am not very familiar with the Norstar family of PBXs
> I've managed to reprogram it for DIDs to each extension, OLI, and will
> risk minor changes to the call-routing, all to get nice integration
> with Asterisk.

the MICS has limitations to the number of DID->extension mappings (I think 
it's in the range of the 30s).  OLI was easy and I use that to track who is 
calling where, but it does no good for telling me where an incoming call 
ended up.  :-)

> All I know for sure is what I've read in the Norstar Installer and
> Coordinator manuals, but that along with some forum posts on
> tek-tips.com lead me to believe it might work without the need for any
> extra equipment for my MICS.

I've read the same posts and have come to a different conclusion.  I have my 
own MICS extension going to * voicemail but the MWI doesn't work (I haven't 
spent a lot of time trying either though) -- I can make it light up with an 
ATA2 and some hookflash magic but I'm trying to keep it all with the single 
PRI instead of hte PRI + channel bank or TDM card (or Sipura) just to get the 
MWI to work.  :-)

-A.
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Re: [Asterisk-Users] Re:How many TDM22P Card can be used on thesame PC ?

On Thursday 18 August 2005 18:08, [EMAIL PROTECTED] wrote:
> Thanks but it doesnt answer my question.

Yes it does -- It answers your question in the sense of "nobody else has done 
it, it seems everyone else goes for a T1+CB, so how about you try it and 
report back since you don't want to take anyone's advice but rather insist 
that it's theoretically possible.

> And nobody knows/mentions (in archives) why it will fail on a
> """reliable""" system.(If you dont know how asterisk handles those cards,
> you can blame everything...)

Actually they have.  Interrupt sharing for one.  Interrupt overhead for 
another.  Drivers which are optimized for minimum latency instead of a 
balance between latency and ability to share interrupts and overhead for a 
third.

But again, we aren't saying "yes it works, this is how" so you pretend not to 
hear.  So since you are obviously the smarter of all of us you can ignore our 
experienced answers and go it your own way.  Just be sure not to bitch about 
the shitty cards/software/users list when you respond after spending the 
money on two solutions...  the one which will theoretically work and the one 
which has been shown to work in practise.

-A.
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Re: [Asterisk-Users] 1-800 number

2005-08-18 Thread dbruce

Isn't that a rather morally bankrupt method of conducting your testing??

TollFree numbers are billed on a per minute basis so someone ends up
paying for the testing you are trying to do...

Try calling your cell phone or your land line. Take responsibility for your
testing rather than passing it on to some unsuspecting third party.

BTW what number are you doing your testing from??? (I want to add it to
my servers blacklist!)

Regards,
Derek

- Original Message -
From: "Matt Hess" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Sent: Thursday, August 18, 2005 2:27 PM
Subject: Re: [Asterisk-Users] 1-800 number


> Just call a milliwatt..?
>
>
> C F wrote:
> > Try your local DMV :)
> >
> > On 8/18/05, Derek Whitten <[EMAIL PROTECTED]> wrote:
> >
> >>try calling comcast.. they are always good for at least 15 minutes of
> >>hold 18778242288
> >>
> >>qworst(qwest) works too.. 1800244
> >>
> >>
> >>On Thu, 2005-08-18 at 06:28, Adam Vocks wrote:
> >>
> >>>Just call tech support for a large company.  Your always on hold longer
> >>>than 10 minutes!
> >>>
> >>>Adam
> >>>
> >>>-Original Message-
> >>>From: [EMAIL PROTECTED]
> >>>[mailto:[EMAIL PROTECTED] On Behalf Of Christoph
> >>>Eicke
> >>>Sent: Thursday, August 18, 2005 8:15 AM
> >>>To: Asterisk Users Mailing List - Non-Commercial Discussion
> >>>Subject: Re: [Asterisk-Users] 1-800 number
> >>>
> >>>I'm trying to solve my Nikotel problem (see previous post) where the
> >>>problem
> >>>is that I get a hangup after 2 minutes, therefore I need some number
> >>>that
> >>>doesn't cost anything and gives me some audio for a long time...
> >>>
> >>>On Thursday 18 August 2005 15:08, Jonathan k. Creasy wrote:
> >>>
> What problem are you trying to solve with this? Just stepping out on a
> limb but it sounds like you are trying to swat a fly with an F-16.
> 
> -Jonathan
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> >>>
> >>>Christoph
> >>>
> Eicke
> Sent: Wednesday, August 17, 2005 4:34 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] 1-800 number
> 
> Hi!
> 
> I'm searching for a 1-800 number that simply plays music for a long
> >>>
> >>>time
> >>>
> (>3mins) and no one picks up. I've bothered the AT&T lines so far when
> trying
> out my SIP->PSTN connection but then always someone answered :-)
> Anyone have a number?
> 
> Christoph
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> >>
> >>--
> >>-BEGIN GEEK CODE BLOCK-
> >>Version: 3.1
> >>GCS/GIT> d-@ s+:+ a? C+++ BLHIS$ U+++ P+> L+++ !E W+++$ N++ o+ K w--
> >>PS+++ PE@ Y+ PGP++ t 5? X !R tv+ b- DI-- D G e+> h r+++ y
> >> --END GEEK CODE BLOCK--
> >>
> >>
> >>-BEGIN PGP SIGNATURE-
> >>Version: GnuPG v1.2.6 (GNU/Linux)
> >>
> >>iD4DBQBDBJMCzGZD3alCWIARAgWjAJj+ulL8T7ejTvoEcEsVZ4nhLpt0AJ9iqD18
> >>KuoT22RWCqHTLzT3DgLvpA==
> >>=lHKS
> >>-END PGP SIGNATURE-
> >>
> >>
> >>___
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> >>
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Re: [Asterisk-Users] XML Revisited - Polycom IP 600

2005-08-18 Thread dbruce

1) Is it possible to create soft-keys?
NO.
2) Is it possible to make the numbers on the key pad select URL's?
NO.
3) Is there a way to dial a phone number by selecting a URL?
Yes. use a "tel:[EMAIL PROTECTED]" url.

Regards,
Derek

- Original Message -
From: <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Sent: Thursday, August 18, 2005 1:04 PM
Subject: Re: [Asterisk-Users] XML Revisited - Polycom IP 600


> I am also looking for some more detailed information on the cababilities
of
> the Polycom 600's microbrowser.  I have almost 100 of the IP600's
deployed,
> but have just now found the time to start seriously experimenting with the
> microbrowser.  I have no problem displaying xhtml and using the
arrow/check
> keys to move around, but would like to be able to do more.
>
> My primary questions are:
> Is it possible to create soft-keys?
> Is it possible to make the numbers on the key pad select URL's?
> Is there a way to dial a phone number by selecting a URL?
>
> I tried using some of the tags that work on mobile phones, such as
> "Accesskey", which allows you to tie the 0-9#* keys to URL's, but so far
> have not been able to find anything that works.  I haven't had a chance to
> try any URL's that might dial a phone number.  Several people on the list
> have mentioned using PHP to make contact directories and agent queues
> available on the phone.  Do your users just use the arrow/check keys to
> navigate or have you found more sophisticated ways?
>
> If anyone has had experience with this, or has suggestions on who to
contact
> about getting more technical information, please let me know!
>
> Eric
>
>
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Re: [Asterisk-Users] Craig R. Saxton/PACE/US is out of the office.

2005-08-18 Thread Kevin P. Fleming


Brian Roy wrote:

  Tell me we're not going to get these until Craig gets back to work 
tomorrow?


No, I've told the list server to stop sending mail to him.
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[Asterisk-Users] Out of G.729 Decoder Licenses!

Hello,

I have SIP and Asterisk. On Asterisk I have 2 liscenses of g729 (from digium
website)
SIP user (100) is calling another SIP user (101).
As 101 is not online, my SIP server is redirecting that call to Asterisk.
Asterisk forward it to 101's voice mail box.

SIP user 100's phone have g729 codec. I havn't buy any codec for SIP server
itself.
But when 100 reach at 101's voice mail, I get this:

Aug 18 18:31:36 WARNING[14511]: codec_g729.c:180 g729tolin_framein: Out of
G.729 Decoder Licenses!

I didn't get it.
Would anybody please explain it.

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Re: [Asterisk-Users] Craig R. Saxton/PACE/US is out of the office.

2005-08-18 Thread Brian Roy


On 8/18/05, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
I will be out of the office starting  08/18/2005 and will not return until08/19/2005.
I will respond to your message when I return.
 
 
Tell me we're not going to get these until Craig gets back to work tomorrow?
 
  
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[Asterisk-Users] asterick and festival...Help!

2005-08-18 Thread John Gruber


Earlier this afternoon I had this working

exten => 2890,1,Answer
exten => 2890,2,GoTo(12)
exten => 2890,12,Wait(1)
exten => 2890,13,Festival('I can say numbers like')
exten => 2890,14,SayNumber(1230001,f)
exten => 2890,15,Wait(1)
exten => 2890,16,HangUp

I was so very proud of myself...

All of a sudden after a reboot I get the following from the same 
call plan


--- (9 headers 0 lines)---
   -- Executing Festival("SIP/1000-2915", "I can say numbers like") in 
new stack

 == Parsing '/etc/asterisk/festival.conf': Found
 == Spawn extension (mytest, 2890, 13) exited non-zero on 'SIP/1000-2915'

and of course the call exits. 


Here is my /etc/asterick/festival.conf

[general]
host=127.0.0.1
port=1314
usecache=no
cachedir=/var/lib/asterisk/festivalcache/
festivalcommand=(tts_textasterisk "%s" 'file)(quit)\n

Everything is running on the same box.  I have rebooted... nothing is 
var log messages either.


The local festival_client connects and I can put in (SayText "I can say 
numbers like") and it works great.


The festival_server log show only this for the calls from asterick:
client(11) Thu Aug 18 17:53:01 2005 : accepted from (my machine name here)
client(11) Thu Aug 18 17:53:01 2005 : disconnected

So it looks like it is connecting right.

How do I put festival in debug or tell asterick to show me the 
"non-zero" return value.


I would not be so painful if I had not bragged about having it 
working  It really was working.


John





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[Asterisk-Users] Craig R. Saxton/PACE/US is out of the office.





I will be out of the office starting  08/18/2005 and will not return until
08/19/2005.

I will respond to your message when I return.

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RE: [Asterisk-Users] VoipJet Problems - anyone?

2005-08-18 Thread Rene Kluwen

This morning, also calls to Mexico were dropped.

Rene Kluwen
Chimit


> No problems with US calls. No calls going through to UK though. My
> account login on the website worked this morning.
>
> Michael
>
>>  Original Message 
>> Subject: RE: [Asterisk-Users] VoipJet Problems - anyone?
>> From: Innocent Evil <[EMAIL PROTECTED]>
>> Date: Thu, August 18, 2005 3:32 pm
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> 
>>
>> I noticed their mysql server is down or can't connect to mysql server.
>> I tried to download there cvs format price list.
>>
>>
>>
>>
>> > -Original Message-
>> > From: [EMAIL PROTECTED]
>> > Sent: Thu, 18 Aug 2005 16:04:30 -0400
>> > To: asterisk-users@lists.digium.com
>> > Subject: [Asterisk-Users] VoipJet Problems - anyone?
>> >
>> > Hi,
>> > Does anyone know what is going on with voipjet?   This
>> > morning/afternoon they just seem to have gone down no word on
>> > their website.
>> > ___
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>> >
>> > http://lists.digium.com/mailman/listinfo/asterisk-users___
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>
>

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Re: [Asterisk-Users] libpri mwi functionality?

2005-08-18 Thread Gary Reuter

On 8/18/05, Andrew Kohlsmith <[EMAIL PROTECTED]> wrote:
> I too am hoping to use these with the norstar MICS but I'm not 100% sure if
> it's that simple; the MICS seems to only want to allow that kind of
> featureset with MCDN (SL1 protocol) and I havne't had enough time to really
> play with that.
> 
I really am not very familiar with the Norstar family of PBXs
I've managed to reprogram it for DIDs to each extension, OLI, and will
risk minor changes to the call-routing, all to get nice integration
with Asterisk.
All I know for sure is what I've read in the Norstar Installer and
Coordinator manuals, but that along with some forum posts on
tek-tips.com lead me to believe it might work without the need for any
extra equipment for my MICS.
I guess I'll only know for sure once i can try it, but for that I need
a Dialplan application like the SrxMWI, but for Digium PRI channels
instead of Sirrix.  Unfortunately, I'm not a good enough programmer to
implement such a command on my own, but I'd be quite able to test,
debug, and even suggest fixes if needed, once the basics functionality
is there.
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RE: [Asterisk-Users] VoipJet Problems - anyone?

2005-08-18 Thread mgraves

No problems with US calls. No calls going through to UK though. My
account login on the website worked this morning.

Michael

>  Original Message 
> Subject: RE: [Asterisk-Users] VoipJet Problems - anyone?
> From: Innocent Evil <[EMAIL PROTECTED]>
> Date: Thu, August 18, 2005 3:32 pm
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> 
> 
> I noticed their mysql server is down or can't connect to mysql server.
> I tried to download there cvs format price list.
> 
> 
> 
> 
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > Sent: Thu, 18 Aug 2005 16:04:30 -0400
> > To: asterisk-users@lists.digium.com
> > Subject: [Asterisk-Users] VoipJet Problems - anyone?
> >
> > Hi,
> > Does anyone know what is going on with voipjet?   This
> > morning/afternoon they just seem to have gone down no word on
> > their website.
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[Asterisk-Users] Re:How many TDM22P Card can be used on thesame PC ?

2005-08-18 Thread kalezade

>>You say that each card generates an interrupt in every 1ms.
>>So for 6 cards, each card needs to be processed atleast in
>>166usec(1ms/6) in their interrupt_routine. Is it teoritically impossible on a
>>perfect configuration/system?


>In theory there isn't much difference between theory and practice.  In 
>practice, there is.
>In other words: you're being advised against this by multiple people.You can 
>heed their advice, or you can spend the $800 and have it fail, and then 
>spend your theoretical $2000 on top of that to get a solution that works, and 
>will work better, and that will scale much better.
>-A.

Thanks but it doesnt answer my question.

>As everyone else has said, you *might* be able to make it work, but it is 
>extremely unlikely. You want to buy a T1 interface and a channel bank.
>Tom

And nobody knows/mentions (in archives) why it will fail on a """reliable""" 
system.(If you dont know how asterisk handles those cards, you can blame
everything...)

Soner Tari wrote:
>Sorry but to have 24 ports you need 6x TDM cards, which adds up to at least 
>$1900. And for 12 ports (3x TDM cards) at least $950. From where do you 
>purchase the TDM cards?

Sorry.. my calculation mistake..

>I have a system with 12 FXs , 8 FXO, and E1 Card working very well.
>Alex

Alex, if you are talking about TDM400P cards, I will go with TDM400P cards.Other
wise I will go with E1+channel bank. 

BDM

- End forwarded message -



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[Asterisk-Users] Craig R. Saxton/PACE/US is out of the office.





I will be out of the office starting  08/18/2005 and will not return until
08/19/2005.

I will respond to your message when I return.

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Re: [Asterisk-Users] Persistent variables disappear when dialing Local extension

2005-08-18 Thread Kevin P. Fleming


Falck Kenneth wrote:


My persistent variables (_XXX or __XXX) don't persist when I dial a
Local extension. I'm doing a forked dial where the other channel is SIP
and the other Local. Is this a known problem? Using Asterisk 1.0.9.


Variable inheritance is a CVS HEAD feature, it is not supported in 1.0.x.
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Re: [Asterisk-Users] Which AGI Development Software is fastest on Asterisk?

2005-08-18 Thread Roger Schreiter


Asterisk schrieb:

I'm looking to develop some custom AGI that will be MySQL intensive.  It
appears Asterisk supports many different development environments.  Which
would be best suited for Asterisk and MySQL?



Hi,

this is the same question as for other applications:
Use C, if you want to optimize runtime, use Perl or PHP
if you need fast developement.

I've developed AGIs in C as well as in Perl. Both languages
are suited well, with its respective advantages and disadvantages.

Others report good experience with PHP, Python, and some are
even using Java.


The interface to asterisk is just stdin and stdout and a little
parsing. So don't worry too much whether your preferred programming
language is suitable for AGI programming!


Roger.


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Re: [Asterisk-Users] static noise with this hardware any advice

2005-08-18 Thread Soner Tari

Since you say TDM04B, I guess you are talking about noise on FXO ports. So, 
my guess is that your opermode setting for zaptel driver is wrong? This may 
be true especially if the noise can be avoided or considerably reduced by 
playing with rx/txgain. At least that was the experience I had with my FXO 
modules. Hope this helps.


- Original Message - 
From: "Patrick Fortin" <[EMAIL PROTECTED]>

To: 
Sent: Thursday, August 18, 2005 10:20 PM
Subject: [Asterisk-Users] static noise with this hardware any advice



Hi

We have static noise problem on our asterisk server. latest stable 
release.

The card is a new TDM04B

We have it installed on the following hardware

Motherboard Intel SE7520BD2SCSI
2x POWER SUPPLY 730W INTEL

I will not mention the other hardware because we have desactivated/changed 
all the other items
The only 2 items that we have not changed is the mobo and the power 
supply.


At first it was on scsi drives but we re-installed using a IDE drive
We deactivated the two onboard nic and tried two different brand.
We have deactivated hyper-treading
We have deactivated USB
We have deactivated SATA
We have tried a noise-cancelling power-bar
We have tried two different phones lines
We have tried several IP phones, Cisco, Snom, Gnet
(There is no noise for a call between two phones)
The phone is connected directly in the nic card so there is no network 
problem possible.

We have tried several TDM Card

Anybody knows if the motherboard or the power-supply could be the problem 
?




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Re: [Asterisk-Users] FW: Asterisk-panel

2005-08-18 Thread Nicolás Gudiño

Hello,On 8/18/05, Soner Tari <[EMAIL PROTECTED]> wrote:
It sounds like web_hostname in your /var/www/html/panel/op_server.cfg is setto your external ip. If you change it to your internal ip, I think you'llhave the opposite of what you describe. I couldn't find a decent solution to
this dilemma. Any one?
In the latest FOP snapshot the web_hostname parameter is optional: you
can comment it out and it will just work from the internal an external
net.

Regards,
-- Nicolás GudiñoBuenos Aires - Argentina
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Re: [Asterisk-Users] SIP message 183 and in band info


Tomá¹ Komárek wrote:
Hello, I have such a problem. I have an * configured as a peer connected 
to the gateway to PSTN.


While calling to the switched off cell phone, the gateway sends to the * 
the SIP message 180 with the SDP part, and also a lot of rtp packets 
containing the operator's in band info.


But * forwards the 180 to the UAC without the sdp part and also without 
the rtp stream.


Is there any way, how to setup the * dialplan to translate all incoming 
180 SIP messages to 183 with the SDP part and also to forward the rtp 
stream to the UAC??


That would be a function of a SIP Proxy, which Asterisk is not.

What is the specific PROBLEM you are experiencing?
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Re: [Asterisk-Users] Voicemail Retrival


Christoph Eicke wrote:


Yes, I do it in the following way. In extension.conf add this line:

exten => ,1,VoiceMailMain(s${CALLERIDNUM})
exten => ,2,Hangup()

Here any extension can call  and then automatically gets directed to their 
voicemail where they have some options.


I hope this helps,


Unless you want to check your voicemail from a phone other than your own 
phone.

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Re: [Asterisk-Users] Does intel 865 board works fine with Asterisk

2005-08-18 Thread steve

On Wed, 17 Aug 2005, jonny hashem wrote:

> 
> I would like to know what are the issues I need to
> look for in a chipset board so I can make sure it
> works fine with digium cards and Asterisk . Is intel
> board 865 fits the description?
> 

Well - I built two systems with Intel Desktop board with 865 chipset.  
They have 3.0GHz HT processor.

I fitted the systems each with a TE405P (gen 1) and two TDM400P boards 
with FXO/FXS.  I looped all the ports and loaded it up to capacity (128 
concurrent calls).  It runs solidly.

The boxes have 1GB ram, and two traditional IDE disks in mirrored setup.

Steve

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[Asterisk-Users] MP3Player cmd issue

2005-08-18 Thread Henry Junior


I am running CVS HEAD (on a Linux-PPC machine.)

My current dialplan generates an error at the console in asterisk  
when I attempt to issue the MP3Player command -- I can't figure out  
why it's not playing the actual audio file?


The rest of the dialplan works great.

Here's what I see in the console:

-- Executing MP3Player("IAX2/[EMAIL PROTECTED]",
"/private/var/lib/asterisk/sounds/audiofile.mp3") in new stack
Aug 18 15:04:19 NOTICE[17387]: chan_iax2.c:3006 iax2_read: I should  
never

be called!
Aug 18 15:04:22 NOTICE[17387]: app_mp3.c:96 timed_read: Poll timed
out/errored out with 0

Does anyone have any ideas what's going on or where I could look to  
trouble shoot this further?


MPG123 plays the audio file fine at the command line, just not thru  
asterisk.  And this same dialplan works fine under CVS HEAD, causing  
MPG123 to play the audio file fine, on OSX so it's something funky  
and I am not sure what it is.

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[Asterisk-Users] Persistent variables disappear when dialing Local extension

2005-08-18 Thread Falck Kenneth

Hi,

My persistent variables (_XXX or __XXX) don't persist when I dial a
Local extension. I'm doing a forked dial where the other channel is SIP
and the other Local. Is this a known problem? Using Asterisk 1.0.9.

My purpose is to fork the original call to two destinations: One going
to an "internal" SIP phone and the other going to an external PSTN
gateway (also using SIP). So the recipient can pick it up from either
phone.

The challenge is that the external PSTN call will be paid for by the
recipient of the original call, and thus it must have a different Caller
ID to make it appear as dialled by him (a requirement). I tried to solve
this by dialing through a Local channel and setting the Caller ID there,
but all my user variables containing info about the call (which are
retrieved from MySQL earlier) are lost.

Thankful for any ideas...

-- 
Kenneth Falck, SWelcom Oy, Ludviginkatu 6-8, 00130 Helsinki, Finland
Private: [EMAIL PROTECTED]  Business: [EMAIL PROTECTED]  GSM: +358405103121
 Never trust an operating system you don't have sources for.

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Re: [Asterisk-Users] Hardware echo cancellation

2005-08-18 Thread ewr


Anybody using hardware echo cancelers a PRI at a time ?

What did you use ? How much did they cost ? Where'd you get em.


After searching for what felt like forever, I finally found a single PRI 
echo canceller.  It is made by ValiantCom, who I believe is based out of 
India.  I installed my first one about 3 weeks ago and so far it is working 
great.  I have over 30 offices on the east coast, and was experiencing 
horrible echo problems on every call to several of our smaller offices.  We 
also experienced what seemed like random cases of occassional echo to other 
numbers.  Since installing the HW canceller I have not had a single case of 
echo.  In addition to voice, we use asterisk to receive faxes over the PRI 
and (as long as the echo canceller is set to unidirectional mode) that also 
continues to work well.  The echo cancellers cost around $1000.  If anyone 
is interested, contact me off-list and I will give you the contact 
information for the US distributor I have been purchasing them through.


I don't mean to breach list etiquette too badly, but on a related subject:
I now have a Tellabs 255D shelf with 16 2572 cards, and a Tellabs 81-8007 
10Amp power supply that I don't need.  I never had the time to hook them all 
up, but as far as I know it all works fine.  I have a punch down block and 
the amphenol cables to go with it, but don't have the wire to connect 
between the power supply and shelf.  I intend to put them on ebay when I 
have a chance, but if anyone is interested now feel free to make an offer.


Eric


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Re: [Asterisk-Users] No translator path exists for channel type MGCP & Comfort noise support incomplete

2005-08-18 Thread kurt turner

the fix is to disallow=g729 in mgcp.conf and to turn off silence supression in the ADIT600kurt turner <[EMAIL PROTECTED]> wrote:

not trashing deb at all.. just wanted to see what would happen with redhat.. I'VE GIVEN UP and I'm reloading DEB.. I'm such a newb at this and I found more doc's with redhat support that's why I wanted to try the switch.. anyways HI HO hi ho back to Deb I go!
know any doc put together for asterisk on debian sarge? especialy with h323 support?Michiel van Baak <[EMAIL PROTECTED]> wrote:
On 09:38, Mon 15 Aug 05, kurt turner wrote:> ONLY ON MONDAY!> > Well it used to work - calls between my aaln's that is. I moved from debain to redhat (same conf. files for asterisk) and this is what I get.. looks like several errors. errors I never got before. Also asterisk isn't observing the digits as I dial them like it used to however it still trys to route the call when I'm finished dialing. Anyone with a though on this?> That's what you get from trashing Debian in favour ofRedHatPlease don't take this message seriously ;) Just couldn'tresist.Sorry-- Michiel van Baakhttp://michiel.vanbaak.info[EMAIL PROTECTED]GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D"Why is it drug addicts and computer afficionados are both called
users?"___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users

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Re: [Asterisk-Users] Polycom SoundPoint 501 power adapter

2005-08-18 Thread Jeremy Gault


Same here on my PolyCom 501.

Also, polarity is positive on the center pin, negative on the outside.

Jeremy Melanson wrote:


All of my 500's and 501's are 12V 400ma as well.

On Thu, 2005-08-18 at 14:00 -0500, Alan Bunch wrote:
 


Mine is 12V 400 ma

alabun

Paul Belanger wrote:

   


Can somebody who has a SoundPoint 501 please confirm the power adapter input / 
output settings:

Input: 120V AC 60HZ 20W
Output: 24V DC 500mA

PB


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--
Jeremy Gault<[EMAIL PROTECTED]>
Network Administrator, WinWorld Corporation
voice: +1.423.473.8084  fax: +1.423.472.9465
fwd: 461771 url: http://www.winworld.cc/

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Re: [Asterisk-Users] Preventing an extension from dialing certainoutbound codes

Benjamin Lawetz wrote:
> Just put 203 in his own context which reacts to those numbers, and then
> include your normal context
> 
> [restrict]
> Exten => _00.,1,goto(unauthorised,1)
> Exten => _087.,1,goto(unauthorised,1)
> Exten => _086.,1,goto(unauthorised,1)
> Exten => unauthorised,1,Playback(invalid)
> Exten => unauthorised,2,wait(2)
> Exten => unauthorised,3,Hangup()
> Include => regular_context
> 

Thanks, I see I have a lot ore learning to do and will get back to
reading my printout of the PDF

Sean

-- 
++
|VOIP: FreeWorldDial 689482 VOIPBuster thecivvie |
|GPG Key: http://thecivvie.fastmail.fm/thecivvie.asc |
++


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Description: S/MIME Cryptographic Signature
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RE: [Asterisk-Users] Which AGI Development Software is fastest onAsterisk?

2005-08-18 Thread Brian C. Fertig

What can you develop in?   What are you comfortable?   I use PHP for
testing
then convert into C shared objects.

..o---o.
Brian Fertig
NOC/Network Engineer
Planet Telecom, Inc.
Tampa, FL Office



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Asterisk
Sent: Thursday, August 18, 2005 4:02 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Which AGI Development Software is fastest
onAsterisk?

I'm looking to develop some custom AGI that will be MySQL intensive.  It
appears Asterisk supports many different development environments.
Which
would be best suited for Asterisk and MySQL?

Bart



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This email was scanned by:  Mcafee GroupShield
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All information provided in this email is considered confidential
and proprietary of Planet Telecom, Inc. and Telecenter Inc.
Use of this information by anyone other than the recipient or 
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Re: [Asterisk-Users] Preventing an extension from dialing certain outbound codes

Andrew Kohlsmith wrote:
> On Thursday 18 August 2005 15:35, Sean Rima wrote:
>> Is there anyway to prevent an extension from dialing certain codes. ie I
>> want to prevent extension 203 from dialing number which start with 00
>> 087 086 etc
> 
> You're thinking about it wrong.
> 
> Devices can only dial #s that match the dialplan in their context.  Simply 
> don't include any extensions that match it.
> 
> It is also painfully obvious that you haven't read much on Asterisk.  I 
> suggest you start by reading the Asterisk Handbook draft, and follow up by 
> poking around the wiki and the mailing list, which is searchable with google.
> 
> http://www.digium.com/handbook-draft.pdf
> http://voip-info.org/
> 
> and include "site: lists.digium.com" in your google terms to search the list.
> 

Ahh I never actually looked at it that way. I am atm reading the
handbook and do browse the wiki site, it is usually my first port of
call. I will read a bit more on the dial plans and contexts and work it out.

-- 
++
|VOIP: FreeWorldDial 689482 VOIPBuster thecivvie |
|GPG Key: http://thecivvie.fastmail.fm/thecivvie.asc |
++


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Re: [Asterisk-Users] Which AGI Development Software is fastest on Asterisk?

2005-08-18 Thread Matthew Boehm


Asterisk wrote:

I'm looking to develop some custom AGI that will be MySQL intensive.  It
appears Asterisk supports many different development environments.  Which
would be best suited for Asterisk and MySQL?

Bart


I use PHP. Love it. Fast, Easy.

-Matthew

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RE: [Asterisk-Users] VoipJet Problems - anyone?

I noticed their mysql server is down or can't connect to mysql server.
I tried to download there cvs format price list.




> -Original Message-
> From: [EMAIL PROTECTED]
> Sent: Thu, 18 Aug 2005 16:04:30 -0400
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] VoipJet Problems - anyone?
>
> Hi,
> Does anyone know what is going on with voipjet?   This
> morning/afternoon they just seem to have gone down no word on
> their website.
> ___
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Re: [Asterisk-Users] VoipJet Problems - anyone?

2005-08-18 Thread Alfredo Manrique

Same here. If you try to login on the website you get a MySQL error but
Asterisk seems to be connected.  I'm guessing their DB is down and
* reads real time from it so it can't find info to work on.

Alfredo.On 8/18/05, Matt <[EMAIL PROTECTED]> wrote:
Hi,Does anyone know what is going on with voipjet?   Thismorning/afternoon they just seem to have gone down no word ontheir website.___Asterisk-Users mailing list
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[Asterisk-Users] Updated Patch to chan_agent.c for PREACKANNOUNCE

2005-08-18 Thread Hadar Pedhazur

First, many thanks to Greg Boehnlein for his patch to chan_agent.c
for adding a "preackannounce" option.

I am running CVS HEAD from 2005/07/31, and the patch failed in a
few hunks, since the code was refactored to add in some CASE
statements where there were compound if statements before.

Anyway, I have successfully updated the patch to work against head
as of 3 weeks ago, and would happily share that with anyone who is
interested (just drop me a line off list).

If a "diff" is preferable to the full 70k of "C", just let me know
what the correct options are for creating a diff suitable for
patching the asterisk tree.

OK, that said, I have a few questions and comments on this topic.
This is my first use of the Queue command (very successfully so
far), but I am afraid that expanding my use will require further
patches, and I would like to verify that first.

1) If I use the syntax:

Member => SIP/100 (rather than member => Agent/100, which maps to
SIP/100)

Then "ackcall" isn't used at all. In other words, a "hard-wired"
member seems to ignore the agents.conf file completely. Is this
the desired behavior? (It isn't for me...)

2) Since agents.conf is a separate file from queues.conf, having
multiple queues does _not_ permit multiple "preackannounce"
messages, each tied to a different queue (this strikes me as
having better been patched into the Queue command). Similarly, you
can't have one queue that has "ackcall=yes", and another with
"ackcall=no".

3) I have the _exact_ same source version of CVS HEAD (from
2005/07/31) running on different servers (after a "cvs co", I tar
the source so that I can be sure I'm running _identical_
versions).

On one machine, when an Agent logs in, I can see it in the DB,
"database show" shows a key of:

//Agents/1001  : [EMAIL PROTECTED];1001

On another machine, the DB shows _nothing_, yet the
AgentCallbackLogin application works correctly (logging agents in
and out), and shows the correct mapping on the CLI during a login.
Still, the DB has _no trace_ of the Agents. I can't explain the
difference in behavior, and would _love_ to have someone solve
that mystery for me.

I'm hoping that I am missing something obvious in the interaction
between the Queue command and the Agents channel, and that some
kind soul here will educate me. Otherwise, I think I might be off
to doing more work in "C" than I ever though I would again in my
life ;-).

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Re: [Asterisk-Users] 1-800 number

2005-08-18 Thread Matt Hess


Just call a milliwatt..?


C F wrote:

Try your local DMV :)

On 8/18/05, Derek Whitten <[EMAIL PROTECTED]> wrote:


try calling comcast.. they are always good for at least 15 minutes of
hold 18778242288

qworst(qwest) works too.. 1800244


On Thu, 2005-08-18 at 06:28, Adam Vocks wrote:


Just call tech support for a large company.  Your always on hold longer
than 10 minutes!

Adam

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Christoph
Eicke
Sent: Thursday, August 18, 2005 8:15 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] 1-800 number

I'm trying to solve my Nikotel problem (see previous post) where the
problem
is that I get a hangup after 2 minutes, therefore I need some number
that
doesn't cost anything and gives me some audio for a long time...

On Thursday 18 August 2005 15:08, Jonathan k. Creasy wrote:


What problem are you trying to solve with this? Just stepping out on a
limb but it sounds like you are trying to swat a fly with an F-16.

-Jonathan

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of


Christoph


Eicke
Sent: Wednesday, August 17, 2005 4:34 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] 1-800 number

Hi!

I'm searching for a 1-800 number that simply plays music for a long


time


(>3mins) and no one picks up. I've bothered the AT&T lines so far when
trying
out my SIP->PSTN connection but then always someone answered :-)
Anyone have a number?

Christoph
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begin:vcard
fn:Matt Hess
n:Hess;Matt
org:LiveWire Networks
adr;dom:;;4577 Pecos St;Denver;CO;80211
email;internet:[EMAIL PROTECTED]
title:Sr. Network Engineer
tel;work:303-458-5667 x 106
tel;fax:303-458-5725
x-mozilla-html:FALSE
url:http://www.livewirenet.com/
version:2.1
end:vcard

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Re: [Asterisk-Users] Polycom SoundPoint 501 power adapter

2005-08-18 Thread Doug Lytle


Paul Belanger wrote:


Can somebody who has a SoundPoint 501 please confirm the power adapter input / 
output settings:

Input: 120V AC 60HZ 20W
Output: 24V DC 500mA


 



Mine is input: 120VAC 60HZ 9W
Output: 12VDC 400mA
Model NO:WD411200400

Doug

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[Asterisk-Users] Craig R. Saxton/PACE/US is out of the office.





I will be out of the office starting  08/18/2005 and will not return until
08/19/2005.

I will respond to your message when I return.

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Re: [Asterisk-Users] VoipJet Problems - anyone?

2005-08-18 Thread Tim Pushor


I am succesfully getting through on their west coast server ...

Tim

Matt wrote:


Hi,
Does anyone know what is going on with voipjet?   This
morning/afternoon they just seem to have gone down no word on
their website.
___
 


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Re: [Asterisk-Users] Which AGI Development Software is fastest on Asterisk?

2005-08-18 Thread Stefan Reuter

On Thu, 2005-08-18 at 13:01 -0700, Asterisk wrote:
> I'm looking to develop some custom AGI that will be MySQL intensive.  It
> appears Asterisk supports many different development environments.  Which
> would be best suited for Asterisk and MySQL?

First you should decide if you want to run short lived AGI script
processes on the same box along with Asterisk or want Asterisk to
connect to one long lived process that serves multiple calls via TCP/IP
(FastAGI). Usually you will prefer FastAGI as it allows AGI scripts to
be processed on a different machine if needed and eliminates the cost of
spawning a process (and additional setup like opening a db connection)
for each call.
Next you want to decide on you favorite programming language, there is
support for almost everything. Have a look at
http://www.voip-info.org/tiki-index.php?page=Asterisk%20AGI
If you want go with Java have a look at Asterisk-Java's support for
FastAGI at http://asterisk-java.sourceforge.net/tutorial.html

=Stefan

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RES: [Asterisk-Users] asterisk seems to load but cannot connect using -r?

2005-08-18 Thread Fábio Sakai

Angus,

Try this command: asterisk -c -r

Fábio Sakai
DGX - Digital Express
Suporte CosmoCall
[EMAIL PROTECTED]
+55 11 3049.8109

-Mensagem original-
De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Em nome de Angus Comber
Enviada em: quinta-feira, 18 de agosto de 2005 16:58
Para: asterisk-users@lists.digium.com
Assunto: [Asterisk-Users] asterisk seems to load but cannot connect using -r?

I installed asterisk on SUSE 9.3.  Stupidly I loaded selected to load 
asterisk from the SUSE DVD - then installed latest asterisk head using cvs. 
At end of asterisk compilation mentioned modules in /modules where from 
another installation.

My telephony cards working ok and if run asterisk just get these warnings:

 [chan_capi.so]Aug 18 20:43:32 WARNING[13248]: loader.c:314 __load_resource: 
/usr/lib/asterisk/modules/chan_capi.so: undefined symbol: ast_smoother_feed
Aug 18 20:43:32 WARNING[13248]: loader.c:543 load_modules: Loading module 
chan_capi.so failed!

Are they serious?

Then I try:
linux:/var/run/asterisk # asterisk -r
Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl 
exist?)
linux:/var/run/asterisk # ls -al
total 5
drwxr-x---   2 asterisk root 112 Aug 18 20:43 .
drwxr-xr-x  13 root root 880 Aug 18 18:44 ..
srwxr-xr-x   1 root root   0 Aug 18 20:43 asterisk.ctl
-rw-r--r--   1 root root   6 Aug 18 20:43 asterisk.pid
linux:/var/run/asterisk #

but  /var/run/asterisk/asterisk.ctl does exit?  how can I fix this?

Is it a problem with those modules in /usr/lib/asterisk/modules?  Should I 
delete them?  What?

Angus


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Re: [Asterisk-Users] Cisco ATA-186 working peer to peer

2005-08-18 Thread Stewart Nelson

Hi Luis,

> Can anyone can tell me if I can connect 2 Cisco ATA-186 in a peer to peer
layout
> (without an Asterisk server registerisng the devices) through Internet?

If running MGCP or SCCP, no.

If running H.323 or SIP, and both ATAs are on static public IPs, no problem.
Just specify the address of each unit as the gateway or proxy for the other.
Disable registration.

If NAT and/or dynamic IP is involved, it depends on what firmware version
you are running, whether the NATs are aware of the protocol being used,
and whether you have administrative control of them.

But, why are you trying to do this?  If you just register the two
units with Free World Dialup or similar, it should work ok with NAT
and dynamic IP, and the config will be provided for you.

--Stewart



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Re: [Asterisk-Users] Preventing an extension from dialing certain outbound codes

On Thursday 18 August 2005 15:35, Sean Rima wrote:
> Is there anyway to prevent an extension from dialing certain codes. ie I
> want to prevent extension 203 from dialing number which start with 00
> 087 086 etc

You're thinking about it wrong.

Devices can only dial #s that match the dialplan in their context.  Simply 
don't include any extensions that match it.

It is also painfully obvious that you haven't read much on Asterisk.  I 
suggest you start by reading the Asterisk Handbook draft, and follow up by 
poking around the wiki and the mailing list, which is searchable with google.

http://www.digium.com/handbook-draft.pdf
http://voip-info.org/

and include "site: lists.digium.com" in your google terms to search the list.

-A.
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RE: [Asterisk-Users] Preventing an extension from dialing certainoutbound codes

2005-08-18 Thread Benjamin Lawetz

Just put 203 in his own context which reacts to those numbers, and then
include your normal context

[restrict]
Exten => _00.,1,goto(unauthorised,1)
Exten => _087.,1,goto(unauthorised,1)
Exten => _086.,1,goto(unauthorised,1)
Exten => unauthorised,1,Playback(invalid)
Exten => unauthorised,2,wait(2)
Exten => unauthorised,3,Hangup()
Include => regular_context

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sean Rima
Sent: August 18, 2005 3:36 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Preventing an extension from dialing
certainoutbound codes

Is there anyway to prevent an extension from dialing certain codes. ie I
want to prevent extension 203 from dialing number which start with 00
087 086 etc

Sean

-- 
++
|VOIP: FreeWorldDial 689482 VOIPBuster thecivvie |
|GPG Key: http://thecivvie.fastmail.fm/thecivvie.asc |
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Re: [Asterisk-Users] 1-800 number

2005-08-18 Thread C F

Try your local DMV :)

On 8/18/05, Derek Whitten <[EMAIL PROTECTED]> wrote:
> try calling comcast.. they are always good for at least 15 minutes of
> hold 18778242288
> 
> qworst(qwest) works too.. 1800244
> 
> 
> On Thu, 2005-08-18 at 06:28, Adam Vocks wrote:
> > Just call tech support for a large company.  Your always on hold longer
> > than 10 minutes!
> >
> > Adam
> >
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of Christoph
> > Eicke
> > Sent: Thursday, August 18, 2005 8:15 AM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [Asterisk-Users] 1-800 number
> >
> > I'm trying to solve my Nikotel problem (see previous post) where the
> > problem
> > is that I get a hangup after 2 minutes, therefore I need some number
> > that
> > doesn't cost anything and gives me some audio for a long time...
> >
> > On Thursday 18 August 2005 15:08, Jonathan k. Creasy wrote:
> > > What problem are you trying to solve with this? Just stepping out on a
> > > limb but it sounds like you are trying to swat a fly with an F-16.
> > >
> > > -Jonathan
> > >
> > > -Original Message-
> > > From: [EMAIL PROTECTED]
> > > [mailto:[EMAIL PROTECTED] On Behalf Of
> > Christoph
> > > Eicke
> > > Sent: Wednesday, August 17, 2005 4:34 AM
> > > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > > Subject: [Asterisk-Users] 1-800 number
> > >
> > > Hi!
> > >
> > > I'm searching for a 1-800 number that simply plays music for a long
> > time
> > >
> > > (>3mins) and no one picks up. I've bothered the AT&T lines so far when
> > > trying
> > > out my SIP->PSTN connection but then always someone answered :-)
> > > Anyone have a number?
> > >
> > > Christoph
> > > ___
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> > > Asterisk-Users@lists.digium.com
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > > To UNSUBSCRIBE or update options visit:
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> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
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[Asterisk-Users] VoipJet Problems - anyone?

2005-08-18 Thread Matt

Hi,
Does anyone know what is going on with voipjet?   This
morning/afternoon they just seem to have gone down no word on
their website.
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[Asterisk-Users] Which AGI Development Software is fastest on Asterisk?

2005-08-18 Thread Asterisk

I'm looking to develop some custom AGI that will be MySQL intensive.  It
appears Asterisk supports many different development environments.  Which
would be best suited for Asterisk and MySQL?

Bart



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[Asterisk-Users] asterisk, Kirk IP600 and Kirk Z-4020

2005-08-18 Thread Vedran Dakic









Hello.

 

Did anyone manage to get these working? 






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[Asterisk-Users] asterisk seems to load but cannot connect using -r ?

2005-08-18 Thread Angus Comber

I installed asterisk on SUSE 9.3.  Stupidly I loaded selected to load 
asterisk from the SUSE DVD - then installed latest asterisk head using cvs. 
At end of asterisk compilation mentioned modules in /modules where from 
another installation.


My telephony cards working ok and if run asterisk just get these warnings:

[chan_capi.so]Aug 18 20:43:32 WARNING[13248]: loader.c:314 __load_resource: 
/usr/lib/asterisk/modules/chan_capi.so: undefined symbol: ast_smoother_feed
Aug 18 20:43:32 WARNING[13248]: loader.c:543 load_modules: Loading module 
chan_capi.so failed!


Are they serious?

Then I try:
linux:/var/run/asterisk # asterisk -r
Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl 
exist?)

linux:/var/run/asterisk # ls -al
total 5
drwxr-x---   2 asterisk root 112 Aug 18 20:43 .
drwxr-xr-x  13 root root 880 Aug 18 18:44 ..
srwxr-xr-x   1 root root   0 Aug 18 20:43 asterisk.ctl
-rw-r--r--   1 root root   6 Aug 18 20:43 asterisk.pid
linux:/var/run/asterisk #

but  /var/run/asterisk/asterisk.ctl does exit?  how can I fix this?

Is it a problem with those modules in /usr/lib/asterisk/modules?  Should I 
delete them?  What?


Angus


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Re: [Asterisk-Users] Polycom SoundPoint 501 power adapter

2005-08-18 Thread Jeremy Melanson

All of my 500's and 501's are 12V 400ma as well.

On Thu, 2005-08-18 at 14:00 -0500, Alan Bunch wrote:
> Mine is 12V 400 ma
> 
> alabun
> 
> Paul Belanger wrote:
> 
> >Can somebody who has a SoundPoint 501 please confirm the power adapter input 
> >/ output settings:
> >
> >Input: 120V AC 60HZ 20W
> >Output: 24V DC 500mA
> >
> >PB
> >
> >
> >___
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> >
> 
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RE: [Asterisk-Users] static noise with this hardware any advice

2005-08-18 Thread Geoff Manning

Patrick Fortin wrote:
> 
> At first it was on scsi drives but we re-installed using a IDE drive
> We deactivated the two onboard nic and tried two different brand.
> We have deactivated hyper-treading
> We have deactivated USB
> We have deactivated SATA
> We have tried a noise-cancelling power-bar
> We have tried two different phones lines
> We have tried several IP phones, Cisco, Snom, Gnet
> (There is no noise for a call between two phones)
> The phone is connected directly in the nic card so there is no network
> problem possible.
> We have tried several TDM Card
> 
> Anybody knows if the motherboard or the power-supply could be the
> problem ? 
> 

I am going through a similar procedure. I contacted Digium support but they
say my hw is not supported (Dell SC420). They are right, but I have another
installation of this in another location without the line noise. AND, I put
the TE110P card in another supported motherboard and have the same noise.

What are your results with zttest?? Is it above 98.5% 

How about IRQ sharing (# lspci -vb ) or (# lspci -v)?

What kernel are you running? I am running 2.6.11 and I am passing this as
kernel parameters at boot:

kernel  /boot/vmlinuz-2.6.11.11 root=/dev/md0 ro acpi=off noapic

The "acpi=off" and "noapic" is what helped get my zttest score to a near
100%.

Hope something here helps, either way I'll be watching this thread to
perhaps solve my own problem.
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RE: [Asterisk-Users] CRM software

2005-08-18 Thread Wiley Siler

Title: CRM software



Go look at the Asterisk @ Home install to see how they got 
Sugar CRM integrated.  It is a good start point and you can build from 
there.
 
W
 


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Lee 
ArcherSent: Thursday, August 18, 2005 8:29 AMTo: Asterisk 
Users Mailing List - Non-Commercial DiscussionSubject: 
[Asterisk-Users] CRM software

Can anyone recommend CRM software with a link into 
Asterisk?  I would like a pop up on caller ID if possible.  I've 
played with the FOP and SugarCRM but can't get them  working 
together.
Regards 
Lee 
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[Asterisk-Users] Preventing an extension from dialing certain outbound codes

Is there anyway to prevent an extension from dialing certain codes. ie I
want to prevent extension 203 from dialing number which start with 00
087 086 etc

Sean

-- 
++
|VOIP: FreeWorldDial 689482 VOIPBuster thecivvie |
|GPG Key: http://thecivvie.fastmail.fm/thecivvie.asc |
++


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[Asterisk-Users] How to get long distance carrier to provide separate billing for several companies that share a PRI to LEC?

2005-08-18 Thread Will Scales

I have a single PRI to our local telco that is shared by
multiple companies.  I would like to find a way to get our
long distance carrier to provide separate billing for these
companies.

I am currently calling Authenticate() to collect an account code
which is written to the call detail record, then I estimate the
charges for each company by parsing the call detail records.
This doesn't work well since we are billed different rates for
intrastate/interstate/international calls and I don't want to
keep up with rates for every NPA-NXX.

Is there a way to pass the account code collected by
Authenticate() to the long distance carrier over PRI?

I can get the long distance carrier to enable account codes
on their switch instead of using Authenticate() in Asterisk,
but this doesn't work because DTMF digits are lost unless
they are dialed *very* slowly.  The DTMF digits are being sent
out-of-band (RFC 2833) by Cisco 79XX phones over SIP.  The long
distance carrier collects the account code after the ISDN SETUP
but before the CONNECT.  There are no problems with DTMF once
a call is connected (so I don't think this is a codec issue).
If I use ChanSpy() to listen to the DTMF being sent to the
l/d carrier I can hear dropped digits unless they are dialed
very slowly.

Any help would be appreciated.

Thanks,
Will Scales
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[Fwd: Re: [Asterisk-Users] Set voicemail maximum length by context]

2005-08-18 Thread Tim Pushor


How embarassing. This was not meant for the list.

My apologies..

Tim

 Original Message 
Subject:Re: [Asterisk-Users] Set voicemail maximum length by context
Date:   Thu, 18 Aug 2005 13:17:15 -0600
From:   Tim Pushor <[EMAIL PROTECTED]>
Reply-To: 	Asterisk Users Mailing List - Non-Commercial Discussion 

To: 	Asterisk Users Mailing List - Non-Commercial Discussion 

References: 	<[EMAIL PROTECTED]> 
<[EMAIL PROTECTED]> <[EMAIL PROTECTED]>




Hey, can you ask mom if she would watch the kids overnight one night 
when Heather gets back? Thats what I'd like to do for her birthday - a 
little getway ...




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Re: [Asterisk-Users] Set voicemail maximum length by context


Mom says she's busy.

Tim Pushor wrote:
Hey, can you ask mom if she would watch the kids overnight one night 
when Heather gets back? Thats what I'd like to do for her birthday - a 
little getway ...


Ben Brown wrote:

Is there any way to set the maximum length of the voicemail based upon 
which context the mailbox is in? I have only found the global setting.


Thanks




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--
Eric Wieling * BTEL Consulting * 504-210-3699 x2120

r: Generate a ringing tone for the calling party, passing no audio from
the called channel(s) until one answers. Use with care and don't insert
this by default into all your dial statements as you are killing call
progress information for the user. Really, you almost certainly do not
want to use this. Asterisk will generate ring tones automatically where
it is appropriate to do so. "r" makes it go the next step and
additionally generate ring tones where it is probably not appropriate to
do so.

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[Asterisk-Users] Directed pickup troubles

2005-08-18 Thread Eric Dantie

I've tried the directed pickup like explained in
http://www.voip-info.org/wiki-PBX+Call+Pickup
but I've got no way doing it work. When I call the ringing phone it use
the second line of that phone

Group call pickup works perfectly.

Is there some special configuration to do to make it works?

Best regards,

Éric
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Re: [Asterisk-Users] Hardware echo cancellation


Alan Bunch wrote:

Anybody using hardware echo cancelers a PRI at a time ?

What did you use ? How much did they cost ? Where'd you get em.

Tell us about them !


I'm playing around with the Tellabs echo canceler.  Nothing to report yet.
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RE: [Asterisk-Users] static noise with this hardware any advice

2005-08-18 Thread Jonathan k. Creasy

The power supply could definitely be the problem. You tried a difference
TDM04B right?
-Jonathan

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Patrick
Fortin
Sent: Thursday, August 18, 2005 3:21 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] static noise with this hardware any advice

Hi

We have static noise problem on our asterisk server. latest stable
release.
The card is a new TDM04B

We have it installed on the following hardware

Motherboard Intel SE7520BD2SCSI
2x POWER SUPPLY 730W INTEL

I will not mention the other hardware because we have
desactivated/changed 
all the other items
The only 2 items that we have not changed is the mobo and the power
supply.

At first it was on scsi drives but we re-installed using a IDE drive
We deactivated the two onboard nic and tried two different brand.
We have deactivated hyper-treading
We have deactivated USB
We have deactivated SATA
We have tried a noise-cancelling power-bar
We have tried two different phones lines
We have tried several IP phones, Cisco, Snom, Gnet
(There is no noise for a call between two phones)
The phone is connected directly in the nic card so there is no network 
problem possible.
We have tried several TDM Card

Anybody knows if the motherboard or the power-supply could be the
problem ?

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Re: [Asterisk-Users] How many TDM22P Card can be used on the same PC ?

2005-08-18 Thread Tom Rymes


Do you mean that you have 5 Digium TDM400P cards installed?

Tom

On Aug 18, 2005, at 1:26 PM, Jonathan k. Creasy wrote:


That's impressive. I have had trouble with more than 2 digium cards in
the same box.

It wasn't even worth messing with, just used a channel bank and a PRI.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alex
Ternero
Sent: Wednesday, August 17, 2005 1:16 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] How many TDM22P Card can be used on the
same PC ?

I have a system with 12 FXs , 8 FXO, and E1 Card working very well.

Alex

-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] nombre de  
Jonathan k.

Creasy
Enviado el: jueves, 18 de agosto de 2005 11:59
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: RE: [Asterisk-Users] How many TDM22P Card can be used on the
same PC ?


No

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Wednesday, August 17, 2005 7:38 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] How many TDM22P Card can be used on the same
PC ?


 Is it possible to use 24 FXS/FXO lines(on 6 PCI slots) at the same  
time

on the
same PC?

I wonder for sound quality and power issues. Can anyone convince me  
that

I
can(not) use 6 TDM22P cards?

Thanks in advance.
BDM.



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RE: [Asterisk-Users] Set voicemail maximum length by context

2005-08-18 Thread Jonathan k. Creasy

Maybe you didn't intend this for the list?

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tim Pushor
Sent: Thursday, August 18, 2005 3:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Set voicemail maximum length by context

Hey, can you ask mom if she would watch the kids overnight one night 
when Heather gets back? Thats what I'd like to do for her birthday - a 
little getway ...

Ben Brown wrote:

> Is there any way to set the maximum length of the voicemail based upon

> which context the mailbox is in? I have only found the global setting.
>
> Thanks
>
>>
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[Asterisk-Users] Craig R. Saxton/PACE/US is out of the office.





I will be out of the office starting  08/18/2005 and will not return until
08/19/2005.

I will respond to your message when I return.

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[Asterisk-Users] static noise with this hardware any advice

2005-08-18 Thread Patrick Fortin


Hi

We have static noise problem on our asterisk server. latest stable release.
The card is a new TDM04B

We have it installed on the following hardware

Motherboard Intel SE7520BD2SCSI
2x POWER SUPPLY 730W INTEL

I will not mention the other hardware because we have desactivated/changed 
all the other items

The only 2 items that we have not changed is the mobo and the power supply.

At first it was on scsi drives but we re-installed using a IDE drive
We deactivated the two onboard nic and tried two different brand.
We have deactivated hyper-treading
We have deactivated USB
We have deactivated SATA
We have tried a noise-cancelling power-bar
We have tried two different phones lines
We have tried several IP phones, Cisco, Snom, Gnet
(There is no noise for a call between two phones)
The phone is connected directly in the nic card so there is no network 
problem possible.

We have tried several TDM Card

Anybody knows if the motherboard or the power-supply could be the problem ?



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Re: [Asterisk-Users] Set voicemail maximum length by context

2005-08-18 Thread Tim Pushor

Hey, can you ask mom if she would watch the kids overnight one night 
when Heather gets back? Thats what I'd like to do for her birthday - a 
little getway ...


Ben Brown wrote:

Is there any way to set the maximum length of the voicemail based upon 
which context the mailbox is in? I have only found the global setting.


Thanks




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[Asterisk-Users] Hardware echo cancellation

2005-08-18 Thread Alan Bunch


Anybody using hardware echo cancelers a PRI at a time ?

What did you use ? How much did they cost ? Where'd you get em.

Tell us about them !

Alan
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Re: [Asterisk-Users] SPA-2100 Analog Telephone Adapter

Eric Wieling aka ManxPower wrote:
> Sean Rima wrote:
>> Does anyone have any experience of these, I have been offered one and am
>> thinking of adding sticking it onto the back of my Asterisk box and just
>> ignore the WAN port if possible, It would be to stick my exisiting
>> phones onto the asterisk box
> 
> No, you would ignore the LAN port.  When I am at home I use this setup:
> 
> Phones -> 2100 FXS ports -> 2100 WAN port -> Ethernet Switch -> Asterisk

Excellent, will get it then

Sean

-- 
++
|VOIP: FreeWorldDial 689482 VOIPBuster thecivvie |
|GPG Key: http://thecivvie.fastmail.fm/thecivvie.asc |
++


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[Asterisk-Users] Set voicemail maximum length by context

2005-08-18 Thread Ben Brown

Is there any way to set the maximum length of the voicemail based upon 
which context the mailbox is in? I have only found the global setting.


Thanks




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RE: [Asterisk-Users] Searching For a Voip Provider

Please change the subject to 'Advertisement of a VoIP Provider'


> -Original Message-
> From: [EMAIL PROTECTED]
> Sent: Thu, 18 Aug 2005 11:55:50 -0700 (PDT)
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] Searching For a Voip Provider
>
> Hi:
>
> Please advice me of a voip provider with reasonable
> reseller program. I was using voipjet and it has a lot
> of problems.
>
> Did anyone experienced asteriskout.com service? They
> have good prices.
>
> Regards;
> Chawki Hammoud
>
>
>
> 
> Start your day with Yahoo! - make it your home page
> http://www.yahoo.com/r/hs
>
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Re: [Asterisk-Users] XML Revisited - Polycom IP 600

2005-08-18 Thread ewr

I am also looking for some more detailed information on the cababilities of 
the Polycom 600's microbrowser.  I have almost 100 of the IP600's deployed, 
but have just now found the time to start seriously experimenting with the 
microbrowser.  I have no problem displaying xhtml and using the arrow/check 
keys to move around, but would like to be able to do more.


My primary questions are:
Is it possible to create soft-keys?
Is it possible to make the numbers on the key pad select URL's?
Is there a way to dial a phone number by selecting a URL?

I tried using some of the tags that work on mobile phones, such as 
"Accesskey", which allows you to tie the 0-9#* keys to URL's, but so far 
have not been able to find anything that works.  I haven't had a chance to 
try any URL's that might dial a phone number.  Several people on the list 
have mentioned using PHP to make contact directories and agent queues 
available on the phone.  Do your users just use the arrow/check keys to 
navigate or have you found more sophisticated ways?


If anyone has had experience with this, or has suggestions on who to contact 
about getting more technical information, please let me know!


Eric 



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Re: [Asterisk-Users] SPA-2100 Analog Telephone Adapter


Sean Rima wrote:

Does anyone have any experience of these, I have been offered one and am
thinking of adding sticking it onto the back of my Asterisk box and just
ignore the WAN port if possible, It would be to stick my exisiting
phones onto the asterisk box


No, you would ignore the LAN port.  When I am at home I use this setup:

Phones -> 2100 FXS ports -> 2100 WAN port -> Ethernet Switch -> Asterisk
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[Asterisk-Users] Festival sounds too wired !!

Hello,

I was just able to connect to my festival server.. but the voice generated
by festival sounds too wired ..really.
I installed only festival, i didn't install speech_tools and couple progams
as  was documented in voip-info.org

How can I tune up festival to have better voice (not as good as like human
speech!)

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Re: [Asterisk-Users] Polycom SoundPoint 501 power adapter

2005-08-18 Thread Alan Bunch


Mine is 12V 400 ma

alabun

Paul Belanger wrote:


Can somebody who has a SoundPoint 501 please confirm the power adapter input / 
output settings:

Input: 120V AC 60HZ 20W
Output: 24V DC 500mA

PB


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[Asterisk-Users] Searching For a Voip Provider

2005-08-18 Thread chawki hammoud

Hi:

Please advice me of a voip provider with reasonable
reseller program. I was using voipjet and it has a lot
of problems.

Did anyone experienced asteriskout.com service? They
have good prices.

Regards;
Chawki Hammoud




Start your day with Yahoo! - make it your home page 
http://www.yahoo.com/r/hs 
 
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Re: [Asterisk-Users] Zaphfc.ko module error

2005-08-18 Thread Remco Barende


Hi!

You didn't state what distro you are running but my guess is that you 
have autoupdate / up2date running. Before the powerfailure the kernel was 
updated and after the powerfailure the box booted the new kernel for which 
you need to recompile the module.


Cheers!
Remco

On Thu, 18 Aug 2005, Terry Wade wrote:


Hi Guys



I have been running a test server for a few days now with * 1.0.9 bristuff
RC8n. I had a power failure and the test machine was not on the ups. When
power was restored I found the following error: FATAL: Error inserting
zaphfc (/lib/modules/2.6.11.4-20a-default/misc/zaphfc.ko): Unknown symbol in
module, or unknown parameter (see dmesg)



My dmesg output:  zaphfc: unsupported module, tainting kernel.



^^
that makes me believe you are now running a newer kernel
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[Asterisk-Users] SPA-2100 Analog Telephone Adapter

Does anyone have any experience of these, I have been offered one and am
thinking of adding sticking it onto the back of my Asterisk box and just
ignore the WAN port if possible, It would be to stick my exisiting
phones onto the asterisk box

Sean

-- 
++
|VOIP: FreeWorldDial 689482 VOIPBuster thecivvie |
|GPG Key: http://thecivvie.fastmail.fm/thecivvie.asc |
++


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[Asterisk-Users] Polycom SoundPoint 501 power adapter

2005-08-18 Thread Paul Belanger

Can somebody who has a SoundPoint 501 please confirm the power adapter input / 
output settings:

Input: 120V AC 60HZ 20W
Output: 24V DC 500mA

PB


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[Asterisk-Users] Craig R. Saxton/PACE/US is out of the office.